Thats sounds interesting too. I will look into that also.
On Wed, Dec 1, 2010 at 1:30 AM, Stefan Schmidt wrote:
> Am 01.12.10 05:10, schrieb Duane Larson:
> > For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
> > Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards
Am 01.12.10 05:10, schrieb Duane Larson:
> For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
> Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards that traffic to
> Asterisk. And since I already have MySQL Cluster working in a redundant
> fashion I am not sure I want
Thank you for the reply.
Comments below...
On 30 Nov 2010 at 20:21, Warren (Warren Selby )
commented about Re: [asterisk-users] Trying to configure a SIP so:
On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz wrote:
I have been told that my logic in extentions.conf is wrong in trying to
confi
For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards that traffic to
Asterisk. And since I already have MySQL Cluster working in a redundant
fashion I am not sure I want to try out MMM MySQL. I do like the idea of
usin
On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz wrote:
> I have been told that my logic in extentions.conf is wrong in trying to
> configure a SIP
> software phone called Express Talk (remote) .
>
> I'd like to make outgoing calls and calls to local extensions.
>
> Could someone please look at my c
I have been told that my logic in extentions.conf is wrong in trying to
configure a SIP
software phone called Express Talk (remote) .
I'd like to make outgoing calls and calls to local extensions.
Could someone please look at my configuration files at
http://pastebin.com/ajp62wqF
and see what
Most of the solutions there are too complex and not really suited for
Asterisk's low usage. I would seriously consider using MySQL MMM and two
MySQL servers in a master-master role. Have the asterisk server also serve
as the MMM-Manager and its not that hard. You have automated failovers in
MySQL i
Hello nice people :)
I have been struggling with trying to get Zaptel from
http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained
from the OpenSolaris Website. I have tried installing all the necessary
packages, yet I keep getting errors no matter if I try using the gcc
availa
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson wrote:
> I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
> High-Availability. I was wondering if it's possible for Asterisk to also
> use multiple database servers for Realtime? Currently with Realtime I am
> only able to
- would you consider a virtual IP and possibly MySQL Multi-Master Manager?
http://mysql-mmm.org/
--
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The Pythian Group - love your data
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Desk: (613) 565-8696 x298
Cell: (613) 266-3763
On Tue, Nov 30, 2010 at 19:45, Mi
The only thing I found workable, is to use a hostname (i.e. asterisk_sql)
and update /etc/hosts according to which SQL server is up or down.
It's a bit of a hassle, and it would be easier if Asterisk supported
fallback SQL servers, but once done it works well.
Mike
From: asterisk-
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime I am
only able to point to a single IP address for a database. If that databas
Sorry never mind!
I got it to work after sof-linking to /lib/, and loading res_jabber.so
first, chan_gtalk.so second.
So in summary:
ln -s /usr/local/lib /lib/
asterisk-cli>modules load res_jabber.so
asterisk-cli>modules load chan_gtalk.so
Cheers!
*José Pablo Méndez
*
--
Hello,
Can't get chan_gtalk.so module to load, neither res_jabber.so:
Asterisk*CLI> module load chan_gtalk.so
Unable to load module chan_gtalk.so
Command 'module load chan_gtalk.so ' failed.
[Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error
loading module 'chan_gtalk.so': l
Hello Steve!
Am Dienstag, den 30.11.2010, 11:49 + schrieb Steve Howes:
> You installed the module, but did you load it in modules.conf?
No, 'cause the modul should be autoloaded, as on Asterisk 1.6 it has
done.
> If you have, unload from CLI and re-load it and see if you get any errors.
I
> I know understand the latency due to the resending .. But if the link was
> have a good speed internet, then resending will make a big latency?
>
> Maybe this latency better than having a cutting voice?
Fundamentally, TCP's congestion-avoidance and loss-recovery logic simply
won't work well w
On Tue, Nov 30, 2010 at 11:25, Carlos Chavez wrote:
>I am having a problem with a Rhino channelbank. I have an Asterisk
> server running 1.6.2.9 and DAHDI 2.4.0 on a CentOS 5.5 system. We have
> a TE420 card with the first port used in E1 mode (R2, 20 channels) and
> the fourth is in T1
On Tue, 30 Nov 2010, Steve Jones wrote:
Just the contrary - Most QoS schemes will drop TCP first, specifically because
it is known that with TCP, the
packet will be resent, so no application will be hurt. UDP is not dropped
first because it is known that there
will likely be more impact.
I
I'm certainly not a network expert (beyond normal network knowledge for an
IT person), and I agree with you TCP SHOULD be dropped first because of what
you said, but I often heard so-called network expert (or at least some
holding jobs as such) say that it`s normal my UDP packets get dropped
becaus
Just the contrary - Most QoS schemes will drop TCP first, specifically
because it is known that with TCP, the packet will be resent, so no
application will be hurt. UDP is not dropped first because it is known that
there will likely be more impact.
I am not aware of any way to run IAX over TCP, a
I am having a problem with a Rhino channelbank. I have an Asterisk
server running 1.6.2.9 and DAHDI 2.4.0 on a CentOS 5.5 system. We have
a TE420 card with the first port used in E1 mode (R2, 20 channels) and
the fourth is in T1 mode for the channelbank. We are using MG2 echo
cancellatio
On Tue, Nov 30, 2010 at 2:28 AM, bilal ghayyad wrote:
> If I ran IAX in TCP port, and in case my network was having a lot of users
> doing browse on the internet and downloading, so in that case and if the IAX
> used TCP port, so the voice will be better than using UDP (because in TCP the
> lo
On Tue, Nov 30, 2010 at 1:00 PM, Steve Totaro
wrote:
> On Tue, Nov 30, 2010 at 4:28 AM, bilal ghayyad wrote:
>> Hi All;
>>
>> Can I run the IAX on TCP port instead of UDP port?
>>
>> If I ran IAX in TCP port, and in case my network was having a lot of users
>> doing browse on the internet and do
On Tue, Nov 30, 2010 at 4:28 AM, bilal ghayyad wrote:
> Hi All;
>
> Can I run the IAX on TCP port instead of UDP port?
>
> If I ran IAX in TCP port, and in case my network was having a lot of users
> doing browse on the internet and downloading, so in that case and if the IAX
> used TCP port, so
> I know understand the latency due to the resending .. But if the link was
have a good speed internet, then resending will make a big latency?
I think the point is that with TCP, congestion will create a vicious circle
of more congestion, while with UDP congestion is bad in itself, but doesn't
Dear;
I know understand the latency due to the resending .. But if the link was have
a good speed internet, then resending will make a big latency?
Maybe this latency better than having a cutting voice?
What if we reduce the packet size and make it TCP, so resending might cause
acceptable del
On Tue, Nov 30, 2010 at 10:17 AM, David Backeberg wrote:
> On Tue, Nov 23, 2010 at 8:25 AM, voip crazy wrote:
>> Hello,
>>
>> I want to analyze the asterisk logs files, looking for all kind of
>> errors, ¿Anyboby knows any asterisk logs analyzer?
>
> You're only going to have the logs for what yo
On Tue, Nov 23, 2010 at 8:25 AM, voip crazy wrote:
> Hello,
>
> I want to analyze the asterisk logs files, looking for all kind of
> errors, ¿Anyboby knows any asterisk logs analyzer?
You're only going to have the logs for what you create logs for.
I create custom logs for the custom things I ne
On Tue, Nov 23, 2010 at 02:25:43PM +0100, voip crazy wrote:
> Hello,
>
> I want to analyze the asterisk logs files, looking for all kind of
> errors, ¿Anyboby knows any asterisk logs analyzer?
less, grep, and friends.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co..
Matt,
We are located on Costa Rica and so far there's just 1 TELCO running the
industrym with the CAFTA treatment the carrier had to open for
interconnection but they get to define the ground rules for the
interconnection.
They are arguing ISDN is and "end customer" circuit and you cannot use it
> On Tue, Nov 30, 2010 at 10:02 AM, Kyle Kienapfel
> wrote:
>> Sounds like they just need to be told its a hilariously bad idea to host
>> anything important on a cellphone.
El 30/11/10 10:16, Mark Deneen escribió:
> But it has a built-in UPS! ;-)
Same as a laptop and far better than a cellpho
I'd suggest you take the opportunity to patch that box while its going to go
offline anyways!
*nix is not immune to security problems despite what some might like to
think! Also, I believe Asterisk has had a couple relatively
serious vulnerabilities that have been patched in 1.4 and 1.6 over the
Just out of curiosity, what country are you in?
I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections. I would ask them for
specifics about the "legal" issues with what you are doing - it sounds to me
like they are just trying
But it has a built-in UPS! ;-)
On Tue, Nov 30, 2010 at 10:02 AM, Kyle Kienapfel wrote:
> Sounds like they just need to be told its a hilariously bad idea to host
> anything important on a cellphone.
>
> On Mon, Nov 29, 2010 at 1:20 PM, Gordon Henderson
> wrote:
>>
>> On Mon, 29 Nov 2010, Gilles
Sounds like they just need to be told its a hilariously bad idea to host
anything important on a cellphone.
On Mon, Nov 29, 2010 at 1:20 PM, Gordon Henderson <
gordon+aster...@drogon.net > wrote:
> On Mon, 29 Nov 2010, Gilles wrote:
>
> > Hello
> >
> > Some SOHO prospects only have a cellphone an
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of
Hello all,
I have a server which is sending INVITEs with a From and Contact header that
contains a domain part of the address (an IP address) that I can't explain.
My sip.conf does not set a domain.
For example in the following line the 123.456.789.012 is the part I can't
explain.
From: "" ;tag=a
On 30 Nov 2010, at 09:47, Michael Nausch wrote:
> I tried to configure Asterisk 1.8 on one of my test-hosts.
>
> I've installed from centos-asterisk.repo
> (http://packages.asterisk.org/centos/$releasever/tested/$basearch/):
>
> [Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No cha
On 30 Nov 2010, at 09:28, bilal ghayyad wrote:
> If I ran IAX in TCP port, and in case my network was having a lot of users
> doing browse on the internet and downloading, so in that case and if the IAX
> used TCP port, so the voice will be better than using UDP (because in TCP the
> lost packet
HI,
I tried to configure Asterisk 1.8 on one of my test-hosts.
I've installed from centos-asterisk.repo
(http://packages.asterisk.org/centos/$releasever/tested/$basearch/):
Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch
Nov 26 15:34:59 Installed: asterisk18-co
Hi All;
Can I run the IAX on TCP port instead of UDP port?
If I ran IAX in TCP port, and in case my network was having a lot of users
doing browse on the internet and downloading, so in that case and if the IAX
used TCP port, so the voice will be better than using UDP (because in TCP the
lost
2010/11/29 Shaun Ruffell
>
>
> Does the "modprobe wctdm24xxp" command complete successfully?
Is there
> anything in dmesg or in the output of dahdi_scan after loading the driver?
>
> Cheers,
> Shaun
>
Hi,
Here is relevant dmesg's output :
[ 13.455729] dahdi: Telephony Interface Registered o
Hi everyone
Does anyone know how to check the TRANSFERRED Target Number is a
local extension or a PSTN number.
Thanks
Nikhil
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